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| 1 /* | 1 /* | 
| 2  * libjingle | 2  * libjingle | 
| 3  * Copyright 2004 Google Inc. | 3  * Copyright 2004 Google Inc. | 
| 4  * | 4  * | 
| 5  * Redistribution and use in source and binary forms, with or without | 5  * Redistribution and use in source and binary forms, with or without | 
| 6  * modification, are permitted provided that the following conditions are met: | 6  * modification, are permitted provided that the following conditions are met: | 
| 7  * | 7  * | 
| 8  *  1. Redistributions of source code must retain the above copyright notice, | 8  *  1. Redistributions of source code must retain the above copyright notice, | 
| 9  *     this list of conditions and the following disclaimer. | 9  *     this list of conditions and the following disclaimer. | 
| 10  *  2. Redistributions in binary form must reproduce the above copyright notice, | 10  *  2. Redistributions in binary form must reproduce the above copyright notice, | 
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| 131 | 131 | 
| 132   VoEWrapper* voe() { return voe_wrapper_.get(); } | 132   VoEWrapper* voe() { return voe_wrapper_.get(); } | 
| 133   int GetLastEngineError(); | 133   int GetLastEngineError(); | 
| 134 | 134 | 
| 135   // Set the external ADM. This can only be called before Init. | 135   // Set the external ADM. This can only be called before Init. | 
| 136   bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); | 136   bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); | 
| 137 | 137 | 
| 138   // Starts AEC dump using existing file. | 138   // Starts AEC dump using existing file. | 
| 139   bool StartAecDump(rtc::PlatformFile file); | 139   bool StartAecDump(rtc::PlatformFile file); | 
| 140 | 140 | 
| 141   // Check whether the supplied trace should be ignored. |  | 
| 142   bool ShouldIgnoreTrace(const std::string& trace); |  | 
| 143 |  | 
| 144   // Create a VoiceEngine Channel. | 141   // Create a VoiceEngine Channel. | 
| 145   int CreateMediaVoiceChannel(); | 142   int CreateMediaVoiceChannel(); | 
| 146 | 143 | 
| 147  private: | 144  private: | 
| 148   typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; | 145   typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; | 
| 149   typedef sigslot:: | 146   typedef sigslot:: | 
| 150       signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; | 147       signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; | 
| 151 | 148 | 
| 152   void Construct(); | 149   void Construct(); | 
| 153   void ConstructCodecs(); | 150   void ConstructCodecs(); | 
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| 436   std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 433   std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 
| 437 | 434 | 
| 438   // Do not lock this on the VoE media processor thread; potential for deadlock | 435   // Do not lock this on the VoE media processor thread; potential for deadlock | 
| 439   // exists. | 436   // exists. | 
| 440   mutable rtc::CriticalSection receive_channels_cs_; | 437   mutable rtc::CriticalSection receive_channels_cs_; | 
| 441 }; | 438 }; | 
| 442 | 439 | 
| 443 }  // namespace cricket | 440 }  // namespace cricket | 
| 444 | 441 | 
| 445 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 442 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 
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